[asterisk-users] asterisk voicemail to email and relaying
Hi list, I'm trying to get some ideas on this subject. Normally astersik sends emails with voicemail attached trough local MTA. As far as i know there is no way for asterisk to authenticate to an external mailserver to relay these emails. Well, these days every provider has some sort of spam blocking, to add to that usually users of asterisk are behid a dynamic IP with no PTR and list grows depending on what target mail server requirements are. Base on these facts i came to conclusion of setting up local MTA to relay emails trough another mail server (another mail server beeing their ISP mail server), i dont have very good results with sendmail/procmail and SASL, its inconsitance, works with some provider not all... I was wonderin what do you guys use for your asterisk boxes? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX machine connect with audiocode SIP device
Am Mittwoch, den 05.09.2007, 22:58 -0700 schrieb satish patel: Dear all I have FAX machine connected with audiocode SIP device i am trying to send fax and when negosiation going on and i start send fax button then my after half page it got stuck in fax machine so is there any codec problem i am useing ulaw/alaw is it fine or not anybody have idea about sending fax with SIP connected device Satish, you already asked twice about fax and asterisk. As far as I can see, no-one answered those questions. Think why that may be: - Because asterisk and fax have been debated often enough? - Because people expect from you to use google instead of pester the mailing list with questions already answered on the web? - Because your mails do not leave the impression that you really tried to achieve things by yourself _first_ and then come answering with a visible amount of experience (indicated by what you tried, why that did not help, and so on)? For your viewing pleasure there are texts about posting questions in mailing lists, like http://www.eyrie.org/~eagle/faqs/questions.html http://perl.plover.com/Questions.html I try not to be overly sarcastic or malevolent, but I could not resist to write this mail. Hope it helps. Anselm PS: Try http://www.google.com/search?q=asterisk+sip+faxbtnI=go ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] alphabetical extension patterns
Am Donnerstag, den 06.09.2007, 10:16 +0530 schrieb Benjamin Jacob:
Hello ppl,
Any way to specify alphabetical exten patterns in the dialplans on Asterisk?
All my users would have alpha/numerical ids. I don't want to add a line
for every user in my dialplans.
I searched around, but couldn't get anything useful. Any way to get
around this?
As from the docs, you can use letters in brackets, like
exten = _[ABC][DEF].,.
From my config I will give you an example of using names for extensions.
In my case, this is only used for incoming external SIP calls, so that
the extensions on my asterisk can be dialled as sip:[EMAIL PROTECTED]
from the internet.
Regular internal extensions are defined in my context [localdialplan],
my Asterisk DB contains several lines like
callroute/names/anselm = 201
callroute/names/flo = 212
8=== extensions.conf
;* Look up exten in database
exten = _...,5,Set(A=${DB(callroute/names/${EXTEN})})
exten = _...,6,GotoIf($[A = A${A}]?900)
exten = _...,7,Goto(localdialplan,${A},1)
exten = _...,900,Congestion()
===8
(you'd need a bit more intelligence for more than one domain, but I
guess that is not what you think of right now)
HTH
Anselm
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Re: [asterisk-users] asterisk voicemail to email and relaying
On Thu, Sep 06, 2007 at 02:07:28AM -0600, Al lists wrote: Hi list, I'm trying to get some ideas on this subject. Normally astersik sends emails with voicemail attached trough local MTA. As far as i know there is no way for asterisk to authenticate to an external mailserver to relay these emails. Sure. Asterisk *never* authenticates. In fact, Asterisk does not handle SMTP in the first place. The MTA does. The MTA may be sendmail, postfix, or even a non-queing MTA like ssmtp. Your question is not an Asterisk question. Well, these days every provider has some sort of spam blocking, to add to that usually users of asterisk are behid a dynamic IP with no PTR and list grows depending on what target mail server requirements are. Base on these facts i came to conclusion of setting up local MTA to relay emails trough another mail server (another mail server beeing their ISP mail server), i dont have very good results with sendmail/procmail and SASL, its inconsitance, works with some provider not all... I was wonderin what do you guys use for your asterisk boxes? procmail is used for delivery to a mailbox. Not for sending. I specifically prefer postfix to sendmail. But that is a matter of taste... For postfix, see http://www.postfix.org/documentation.html and also specifically http://www.postfix.org/SASL_README.html and http://www.postfix.org/TLS_README.html . The latter is not well-written, though. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] bridge on DIVA card and how to see it
In the capi.conf file, there is a bridge option that allow to native bridging (CAPI line interconnect) if available, and I found this in the capi-user mailing list : I suggest you put bridge=yes into each interface. Then, when Asterisk bridges two channels, it looks for the possibility to do a native bridge (call the bridge code of the channel module). In case of SIP (when reinvite=yes is set), the SIP phones are set to send the voice data directly to the other phone and not bother Asterisk with that voice data. It is an interesting feature, and I put the right value in the conf file; but how to see the effect of this parameter ? All my test show that when asterisk run out of steam, the isdn calls too. Does this parameter function really ? How can I perform my test in order to ascertain it ? P.S. : I used cpufreq in order to curb the cpu, and adjust the load with a kernel compilation (with j parameter). _ Découvrez le Blog heroic Fantaisy d'Eragon! http://eragon-heroic-fantasy.spaces.live.com/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk voicemail to email and relaying
Am Donnerstag, den 06.09.2007, 02:07 -0600 schrieb Al lists: Hi list, I'm trying to get some ideas on this subject. Normally astersik sends emails with voicemail attached trough local MTA. As far as i know there is no way for asterisk to authenticate to an external mailserver to relay these emails. Well, these days every provider has some sort of spam blocking, to add to that usually users of asterisk are behid a dynamic IP with no PTR and list grows depending on what target mail server requirements are. Base on these facts i came to conclusion of setting up local MTA to relay emails trough another mail server (another mail server beeing their ISP mail server), i dont have very good results with sendmail/procmail and SASL, its inconsitance, works with some provider not all... I was wonderin what do you guys use for your asterisk boxes? I have good experience with exim4, the default config needs some tweaking (at least under Debian) for SSL and AUTH stuff, but that is fairly documented and not difficult to setup. I only have one upstream provider, a so-called smarthost, so I need not fear it will break with any other mail host. YMMV. Of course running exim4 only for mail-forwarding is a bit like hunting sparrows with cannons (or whatever the equivalent english phrase is :-) but then, it gets the job done, and without any mail in the queue its memory footprint and cpu usage are neglible. BR Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX machine connect with audiocode SIP device
Thank for suggestion now i have done it and it is working fine One thing i have find many document but i was confuse thats why i have put it on mailing list if u have or anybody have problem then i m sorry for that. Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Mittwoch, den 05.09.2007, 22:58 -0700 schrieb satish patel: Dear all I have FAX machine connected with audiocode SIP device i am trying to send fax and when negosiation going on and i start send fax button then my after half page it got stuck in fax machine so is there any codec problem i am useing ulaw/alaw is it fine or not anybody have idea about sending fax with SIP connected device Satish, you already asked twice about fax and asterisk. As far as I can see, no-one answered those questions. Think why that may be: - Because asterisk and fax have been debated often enough? - Because people expect from you to use google instead of pester the mailing list with questions already answered on the web? - Because your mails do not leave the impression that you really tried to achieve things by yourself _first_ and then come answering with a visible amount of experience (indicated by what you tried, why that did not help, and so on)? For your viewing pleasure there are texts about posting questions in mailing lists, like http://www.eyrie.org/~eagle/faqs/questions.html http://perl.plover.com/Questions.html I try not to be overly sarcastic or malevolent, but I could not resist to write this mail. Hope it helps. Anselm PS: Try http://www.google.com/search?q=asterisk+sip+faxbtnI=go ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk on UML (User Mode Linux)
What's the current thinking on running Asterisk in a UML environment? I saw some discussion about Xen and asterisk on a Xen DomU. I'm currently running Asterisk in a UML and have noticed poorer quality on calls. I'm only using SIP and IAX2 trunks. No hardware adapters. I guess timing is important, but even if I could get the provider to install a kernel with the Zaptel Dummy timing device compiled in (impossible to install kernel modules in UML), I'm not convinced this would necessarily provide an accurate enough timing device. Is anyone else running their Asterisk instance in UML? If anyone is, what's the preferred way to keep timing accurate? Thinking I may have been too hasty in switching to UML... S. -- Simon Tennant ___ http://imaginator.com/~simon/contact signature.asc Description: OpenPGP digital signature ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bridge on DIVA card and how to see it
On Thu, 6 Sep 2007, lemmel lemmel wrote: In the capi.conf file, there is a bridge option that allow to native bridging (CAPI line interconnect) if available, and I found this in the capi-user mailing list : I suggest you put bridge=yes into each interface. Then, when Asterisk bridges two channels, it looks for the possibility to do a native bridge (call the bridge code of the channel module). In case of SIP (when reinvite=yes is set), the SIP phones are set to send the voice data directly to the other phone and not bother Asterisk with that voice data. It is an interesting feature, and I put the right value in the conf file; but how to see the effect of this parameter ? All my test show that when asterisk run out of steam, the isdn calls too. Does this parameter function really ? How can I perform my test in order to ascertain it ? This function does work well. But it works if your ISDN card/driver supports it only. If you have a DIVA Server card, then you can use it. The bridge is done on the DIVA cards DSPs without CPU power. There are three possibilities to see if it really is working: 1) when you type 'capi show channels', you should see a 'G' (for bridGed) in the isdnstate column. 2) Use 'set verbose 5' and 'capi debug' to see the CAPI command when the call is activated. There should be some FACILITY_REQs and infos like 'Line Interconnect activated'. 3) Use 'set verbose 9' and 'capi debug' to see even all Voice Data as CAPI commands. If the bridge is active, there shouldn't be any DATA_B3 commands any more. Of course, this only works if both channels are CAPI and both controllers supports that. Also, if you have allow= set in your capi.conf (use of RTP with DIVA), the Line-Interconnect may not be activated (if so, please contact me). Armin ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk voicemail to email and relaying
Anselm Martin Hoffmeister wrote: Am Donnerstag, den 06.09.2007, 02:07 -0600 schrieb Al lists: Hi list, I'm trying to get some ideas on this subject. Normally astersik sends emails with voicemail attached trough local MTA. As far as i know there is no way for asterisk to authenticate to an external mailserver to relay these emails. Well, these days every provider has some sort of spam blocking, to add to that usually users of asterisk are behid a dynamic IP with no PTR and list grows depending on what target mail server requirements are. Base on these facts i came to conclusion of setting up local MTA to relay emails trough another mail server (another mail server beeing their ISP mail server), i dont have very good results with sendmail/procmail and SASL, its inconsitance, works with some provider not all... I was wonderin what do you guys use for your asterisk boxes? I have good experience with exim4, the default config needs some tweaking (at least under Debian) for SSL and AUTH stuff, but that is fairly documented and not difficult to setup. I only have one upstream provider, a so-called smarthost, so I need not fear it will break with any other mail host. YMMV. Of course running exim4 only for mail-forwarding is a bit like hunting sparrows with cannons (or whatever the equivalent english phrase is :-) but then, it gets the job done, and without any mail in the queue its memory footprint and cpu usage are neglible. BR Anselm I use http://www.dnsexit.com/Direct.sv?cmd=mailRelay to get around port 25 blockage at home and also avoid going into the spam blackhole. It has an option for no authentication if coming from a defined IP address. That gets around setting up any kind of smarthost authentication. Just make sure your box is not an open relay. Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy sound while converting alaw to ulaw
Benoit Panizzon wrote: Hi there I europe alaw is usual. I have a SIP Phone which perferes ulaw. When my * box has to transcode alaw to ulaw the sound get's one way choppy. (alaw = ulaw is choppy, ulaw = alaw is fine). I managed to fix the issue by forcing my SIP phone to use alaw only, but is this a know issue with asterisk 1.2.13? -Benoit- I do not believe that there is an issue in Asterisk. Is this a heavily used box? What does top show when making the call that is choppy. When you say your phone prefers it, do mean alaw is listed before ulaw? It that is the case, then it does not prefer it, it just came that way, default from the factory. Anyways, I find if bandwidth (and that is not even the case for ulaw/alaw) is a problem then transcode. If you do not have to transcode, use the same codec end to end. Then it is just passing data on the wire and not CPU intensive. Thanks, Steve ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multitenant or Multiple virtual machines
Hi all, We want to offer hosted PBX services to some of our clients (maybe 10-20) and were wondering if it makes sense to get a software package capable of handling multiple virtual tenants or if we should just create multiple virtual machines in our server each running a single- tenant license of the software. We have been researching virtual PBX software for asterisk for a couple of weeks and the number of solutions that we found that can handle multi-tenant needs are limited and even the ones available can not do everything some of our clients need. On the other side, there is a large quantity of single-tenant packages out there which seem more feature-complete than the multi-tenant versions we have found. Since we're not going to be doing any transcoding and using only SIP (no IAX or ZAP channels), we started pondering about the virtual machine solution (small number of extensions and simultaneous calls; we don't expect the number of simultaneous calls to exceed 50). Would you guys recommend it? The only thing disadvantage we have thought of so far is that when a client happens to call a number that is hosted by one of the other clients, the call may end up going up to the SIP carrier and back down to us, unless we carefully setup something like DUNDi, which we have no experience with and we don't know if these single-tenant packages even handle DUNDi setup thru their web management interface. Any opinions/comments/recommendations? Before anyone recommends just buying the virtual PBX service from someone else, we _really_ want to do this in-house :) Thanks ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multitenant or Multiple virtual machines
[EMAIL PROTECTED] wrote: Hi all, We want to offer hosted PBX services to some of our clients (maybe 10-20) and were wondering if it makes sense to get a software package capable of handling multiple virtual tenants or if we should just create multiple virtual machines in our server each running a single- tenant license of the software. We have been researching virtual PBX software for asterisk for a couple of weeks and the number of solutions that we found that can handle multi-tenant needs are limited and even the ones available can not do everything some of our clients need. On the other side, there is a large quantity of single-tenant packages out there which seem more feature-complete than the multi-tenant versions we have found. Since we're not going to be doing any transcoding and using only SIP (no IAX or ZAP channels), we started pondering about the virtual machine solution (small number of extensions and simultaneous calls; we don't expect the number of simultaneous calls to exceed 50). Would you guys recommend it? The only thing disadvantage we have thought of so far is that when a client happens to call a number that is hosted by one of the other clients, the call may end up going up to the SIP carrier and back down to us, unless we carefully setup something like DUNDi, which we have no experience with and we don't know if these single-tenant packages even handle DUNDi setup thru their web management interface. Any opinions/comments/recommendations? Before anyone recommends just buying the virtual PBX service from someone else, we _really_ want to do this in-house :) Thanks Then try it in-house. It should not take too long to setup and use SIPP to test. It sounds like you have already made up your mind, so stop wasting time and try it. Then post your results back to the list. Since you do not list the shortcomings of the products you have looked at, I am afraid that making any recommendations is impossible. Thanks Steve ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multitenant or Multiple virtual machines
Thanks for the prompt response. I apologize if my message came the wrong way. The objective of my message was to know whether anyone used multiple instances of asterisk (10-20) within virtual machines and how well it behaves under that scenario. I know there are many people using single-tenant and multi-tenant versions of asterisk management and billing packages, but I don't really know if anyone is using it within virtual machines and how well that scales. I wouldn't mind trying it, but since it will probably involve having to purchase multiple software licenses, we want to avoid the $ expense (although we don't mind spending our time researching and testing) if it may not work properly. Thanks again On Sep 6, 2007, at 8:04 AM, Steve Totaro wrote: [EMAIL PROTECTED] wrote: Hi all, We want to offer hosted PBX services to some of our clients (maybe 10-20) and were wondering if it makes sense to get a software package capable of handling multiple virtual tenants or if we should just create multiple virtual machines in our server each running a single- tenant license of the software. We have been researching virtual PBX software for asterisk for a couple of weeks and the number of solutions that we found that can handle multi-tenant needs are limited and even the ones available can not do everything some of our clients need. On the other side, there is a large quantity of single-tenant packages out there which seem more feature-complete than the multi-tenant versions we have found. Since we're not going to be doing any transcoding and using only SIP (no IAX or ZAP channels), we started pondering about the virtual machine solution (small number of extensions and simultaneous calls; we don't expect the number of simultaneous calls to exceed 50). Would you guys recommend it? The only thing disadvantage we have thought of so far is that when a client happens to call a number that is hosted by one of the other clients, the call may end up going up to the SIP carrier and back down to us, unless we carefully setup something like DUNDi, which we have no experience with and we don't know if these single-tenant packages even handle DUNDi setup thru their web management interface. Any opinions/comments/recommendations? Before anyone recommends just buying the virtual PBX service from someone else, we _really_ want to do this in-house :) Thanks Then try it in-house. It should not take too long to setup and use SIPP to test. It sounds like you have already made up your mind, so stop wasting time and try it. Then post your results back to the list. Since you do not list the shortcomings of the products you have looked at, I am afraid that making any recommendations is impossible. Thanks Steve ___ Sign up now for AstriCon 2007! September 25-28th. http:// www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk voicemail to email and relaying
On Thu, Sep 06, 2007 at 07:30:57AM -0400, Steve Totaro wrote: I use http://www.dnsexit.com/Direct.sv?cmd=mailRelay to get around port 25 blockage at home and also avoid going into the spam blackhole. It has an option for no authentication if coming from a defined IP address. That gets around setting up any kind of smarthost authentication. More and more providers require authentication of some sort. Some have switched to use the submission port (587) for local systems. Just make sure your box is not an open relay. There's a very simple way to do that: only handle mail from the machine itself. Unless you do need to set up a proper mail server for the whole network or whatever. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bridge on DIVA card and how to see it
Thanks for your quick answer :-).
I am a rookie in all this telephony problem, so I'll try to be verbose.
This function does work well. But it works if your ISDN card/driver
supports
it only.
I currently have a Diva Server 4BRI Rev 2, and it seems that there is DSP on
it (voice detection, and so on), so it is correct ?!
There are three possibilities to see if it really is working:
1) when you type 'capi show channels', you should see a 'G' (for bridGed)
in
the isdnstate column.
I just perform this check, and I didn't see the G.
I have two S0 connected to a PBX (Siemens) and on the other side I have my
Diva card.
This is my capi.conf :
*
[contr1]
isdnmode=did
incomingmsn=*
controller=1
group=1
relaxdtmf=on
faxdetect=off
accountcode=
context=toto
echocancelold=yes
devices=4
bridge=yes
*
and I putted the 2 adapters to one M-Adapter.
When I dialed 122 (on the first real adapter), I heard a voice asking for
number to dialed, and I give the 103. When the call is established, I had :
*
Line-Name NTmode state i/o bproto isdnstate ton number
contr1#04noConn I trans *BS 0x00 '107'-'122'
contr1#03noConn O trans *BPS0x00 'b'-'103'
contr1#02noDisc - trans 0x00 ''-''
contr1#01no- - trans 0x00 ''-''
*
When I made a blind transfert to the 104, I had :
Line-Name NTmode state i/o bproto isdnstate ton number
*
contr1#04noDisc - trans 0x00 ''-''
contr1#03noConn O trans *BPS0x00 'b'-'103'
contr1#02noConn O trans *BPS0x00 'b'-'104'
contr1#01no- - trans 0x00 ''-''
*
and finally, when I dial an extension, I do :
exten = _10X,n,Dial(CAPI/contr1/b:${EXTEN}||tT)
Also, if you have allow= set in your capi.conf (use of RTP with DIVA), the
Line-Interconnect may not be activated (if so, please contact me).
I didn't see this parameter in my capi.conf and in the capi.conf example, is
it some specific DIVA parameter ? Where can I found those parameter, in Diva
doc ?
P.S. :
linux : 2.6.21
Diva server drivers : 8.3-107-83 (the latest ones)
asterisk : 1.4.8
chan_capi : 1.0.1
_
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Re: [asterisk-users] DTMF Relay Problems
Thanks. Will check that out. Joseph -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Wednesday, September 05, 2007 2:38 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DTMF Relay Problems In article [EMAIL PROTECTED], Joseph Begumisa [EMAIL PROTECTED] wrote: I have a client setup where a T1 is terminated into a Cisco IAD2430 Series device which then interfaces with a Digium Wildcard TE110P card in a server running Asterisk 1.2.23. I am having a problem with the DTMF tones being passed to the Asterisk server. Wrong tones are being passed to the server especially during the digital receptionist menu selections. Setting relaxdtmf=yes does not seem to address the situation. Any pointers? Try the patch at http://bugs.digium.com/view.php?id=10535 and see if it helps. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk
On Sat, 1 Sep 2007, Jay R. Ashworth wrote: On Sun, Sep 02, 2007 at 04:38:19AM +0300, Tzafrir Cohen wrote: You mentioned that the two disks are identical. Hence there's a large chance that they're from the same batch. This increases the chance of them failing together :-p In practice, though, we've never had both halves of a RAID 1 pair fail in the same month. I've had 3 drives in a 6-drive unit fail in a 36-hour period )-: It was a Dell. 3 x WDC drives, 3 x Seagate. It was the Seagate drives that failled IIRC. Gordon ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 31 seconds because it is directly bridged to another RTP stream
Hi list, I have a problem with 2 or 3 specific clients. In the 6 minute, the voip client hear the other one, but the other side can't hear. After 30 seconds, the both sides recover the audio. And in the asterisk i have the next notice will not be disconnected in 31 seconds because it is directly bridged to another RTP stream Any comments. Regards Guillermo Rodriguez Songomem-Blutu ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bridge on DIVA card and how to see it
On Thu, 6 Sep 2007, lemmel lemmel wrote: Thanks for your quick answer :-). I am a rookie in all this telephony problem, so I'll try to be verbose. This function does work well. But it works if your ISDN card/driver supports it only. I currently have a Diva Server 4BRI Rev 2, and it seems that there is DSP on it (voice detection, and so on), so it is correct ?! Yes. This card supports it. There are three possibilities to see if it really is working: 1) when you type 'capi show channels', you should see a 'G' (for bridGed) in the isdnstate column. I just perform this check, and I didn't see the G. I have two S0 connected to a PBX (Siemens) and on the other side I have my Diva card. This is my capi.conf : * [contr1] isdnmode=did incomingmsn=* controller=1 group=1 relaxdtmf=on faxdetect=off accountcode= context=toto echocancelold=yes devices=4 bridge=yes * This is not quite correct. Your card actually has 4 controllers, so you need to create 4 sections (contr1, contr2, ...) with devices=2 each. !Oh, you use M-Adapter. Then your 4 channels should be working! Also, to make use of the DSPs, don't set softdtmf/relaxdtmf. And since you are using a newer driver, use echocancel=yes and leave echocancelold=off, otherwise your echo-canceler will not work. and I putted the 2 adapters to one M-Adapter. When I dialed 122 (on the first real adapter), I heard a voice asking for number to dialed, and I give the 103. When the call is established, I had : * Line-Name NTmode state i/o bproto isdnstate ton number contr1#04noConn I trans *BS 0x00 '107'-'122' contr1#03noConn O trans *BPS0x00 'b'-'103' contr1#02noDisc - trans 0x00 ''-'' contr1#01no- - trans 0x00 ''-'' * When I made a blind transfert to the 104, I had : Line-Name NTmode state i/o bproto isdnstate ton number * contr1#04noDisc - trans 0x00 ''-'' contr1#03noConn O trans *BPS0x00 'b'-'103' contr1#02noConn O trans *BPS0x00 'b'-'104' contr1#01no- - trans 0x00 ''-'' * Okay, both should be bridged. So somehow it is not activated. So please provide a log with set verbose 5 capi debug Also, if you have allow= set in your capi.conf (use of RTP with DIVA), the Line-Interconnect may not be activated (if so, please contact me). I didn't see this parameter in my capi.conf and in the capi.conf example, is it some specific DIVA parameter ? Where can I found those parameter, in Diva doc ? It is a capi.conf (chan-capi) parameter and is part of the example provided by chan-capi package. So far the DIVA Server cards are the only cards which can do RTP. But it is okay to leave it off. Armin ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Debugging to separate log file
Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? Thank you! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY 14624 Office: 888-865-0065 Ext. 202 Mobile: (585) 721-8679 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sysmaster and Asterisk
Hello Guys, I am unable to make calls to outside number from some of my extensions. Internally I am able to make and receive calls between extensions and also I am able to receive call from outside number. Any suggestions? Then in am thinking of getting rid of Sysmaster and configure Trixbox to do the entire job that currently my Sysmaster is doing. Any suggestions..? Mani ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Debugging to separate log file
On Thu, 2007-09-06 at 09:58 -0400, Jason Martin wrote: Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? I always find it easier to extract the SIP messaging traffic by using tcpdump or ngrep. If you use tcpdump, you can always pass the traffic through ngrep later, as well as passing it through Wireshark to get the pretty SIP traffic graphs, etc. -- Jared Smith Community Relations Manager Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rxfax() problem - fax signal seems to be ignored
[RESOLVED]
Hello Andrew and thx you for your response, which led me to the solution.
You are right concerning the Ringing() and Answer(), so I put this out of my
dialplan.
The way to test with a std phone is a good idea, and permit me to hear the
spandsp CED tone. Very easy to do, and I'm still asking myself why I did'nt
though to that before :-)
This led me to the conclusion that the ||debug was in cause : when using it,
rxfax() answer a coming fax with a CNG tone instead of a CED tone.
I simply removed this ||debug argument, and now when I call the 300
extension, I can hear the CED tone like expected.
Using ||debug acts like if I had used |caller instead. Seems to be a bug.
[/RESOLVED]
2007/9/5, Andrew Joakimsen [EMAIL PROTECTED]:
On 9/5/07, Pirlouwi [EMAIL PROTECTED] wrote:
Hello,
my configuration is the following:
a TDM400P board with an fxs and fxo daughter boards on it.
I thus connect a fax to my FXS port, after having verified that this
port
was correctly functioning. For this, I had tried before with a simple
phone,
and with some basic voicemail exten scripts.
Here is my simple dialplan for my fax reception:
exten = 300,1,Ringing()
exten = 300,n,Answer()
exten = 300,n,Set(FAXFILE=/tmp/test.tif)
exten = 300,n,rxfax(${FAXFILE}||debug)
Why? exten = 300,1,rxfax(/tmp/test.tif||debug) would do the same
exact thing. No need to indicate ringing and no need to answer the
call. Besides that it is just incorrect you are never going to have
correct answer supervision on an analog line, so don't even try.
I then dialed 300 on my fax machine, and expected to be lucky and to
obtain
a /tmp/test.tif file after faxing completion.
But instead, I always got such error in the /var/log/asterisk/full log
file:
What if you just use a regular analog phone and dial 300? What
happens? What if you remove the ||Debug from your RxFax dialstring?
[Sep 5 13:42:24] DEBUG[1298] pbx.c: Launching 'Ringing'
[Sep 5 13:42:24] DEBUG[1298] chan_zap.c: Took Zap/1-1 off hook
[Sep 5 13:42:24] DEBUG[1298] pbx.c: Launching 'Set'
[Sep 5 13:42:24] VERBOSE[1298] logger.c: -- Executing [
[EMAIL PROTECTED]:3]
Set(Zap/1-1, FAXFILE=/tmp/test.tif) in new stack
[Sep 5 13:42:24] DEBUG[1298] pbx.c: Launching 'RxFAX'
Notice how your own logs prove that 0ms elapse between the time you
incorrectly indicate ringing on the channel and the time RxFax begins.
I have enabled the #define LOG_FAX_AUDIO inside spandsp library, and two
audio files (fax-rx-audio-b7933500-070905134224 and
fax-tx-audio-b7933500-070905134224) appeared in /tmp.
Just listen into the line. When you execute RxFax it will play fax
tones just as if another faxmachine answered -- not CNG tones
This is not the case in my setup. What did I wrong?
Thx for your help.
What version of Linux, Asterisk, Zaptel, SpanDSP app_rxfax are you
using?
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[asterisk-users] (txfax+spandsp) fax is successfully sent, but Asterisk keeps sending DelayedRetries.
Hello my current test is to send a fax to an analog fax (# is 5656) using the Asterisk spooler. I created this file which I copied into /var/spool/asterisk/outgoing directory: # fax_out.call Channel: Zap/4/5656 MaxRetries: 50 RetryTime: 60 WaitTime: 45 Application: txfax Data: /tmp/test.tif|caller The fax was successfully sent, immediately. But after this successfully sending, Asterisk keeps retrying to send this file. The fax_out.call file do not disappear from the spool directory, but some messages are added to the end of the .call file, like this: Channel: Zap/4/5656 MaxRetries: 50 RetryTime: 60 WaitTime: 45 Application: txfax Data: /tmp/test.tif|caller StartRetry: 13740 1 (1189086695) DelayedRetry: 13740 0 (1189086756) DelayedRetry: 13740 0 (1189086817) Any idea how to correct that? The fact that the call file is removed from the spool/outgoing directory, does it automatically means that the sending was successful? _Pirlouwi ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bridge on DIVA card and how to see it
So please provide a log with set verbose 5 capi debug Well, let's go for the debug log :-). In this capture, I used only one adapter, and I performed a call to asterisk, which dialed a number (as described in the previous mail) ; so this is the log : -- Executing [EMAIL PROTECTED]:2] Dial(CAPI/contr1#02/123-7, CAPI/contr1/b:103||tT) in new stack data = contr1/b:103 format=8 parsed dialstring: 'contr1' 'b' '103' '' capi request controller = 1 == contr1#01: setting format alaw - 0x8 (alaw) parsed dialstring: 'contr1' 'b' '103' '' capi: peerlink -1 allocated, peer is unlinked == contr1#01: Call CAPI/contr1#01/103-8 (pres=0x03, ton=0x00) CONNECT_REQ ID=002 #0x0753 LEN=0047 Controller/PLCI/NCCI= 0x1 CIPValue= 0x1 CalledPartyNumber = 80103 CallingPartyNumber = 00 83b CalledPartySubaddress = default CallingPartySubaddress = default BProtocol B1protocol = 0x1 B2protocol = 0x1 B3protocol = 0x0 B1configuration= default B2configuration= default B3configuration= default GlobalConfiguration= default BC = default LLC = default HLC = default AdditionalInfo BChannelinformation= 00 00 Keypadfacility = default Useruserdata = default Facilitydataarray = default SendingComplete= default -- Called contr1/b:103 CAPI devicestate requested for contr1#01/103 CONNECT_CONF ID=002 #0x0753 LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 -- contr1#01: received CONNECT_CONF PLCI = 0x101 INFO_IND ID=002 #0x08ca LEN=0017 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x1e InfoElement = 81 88 INFO_RESP ID=002 #0x08ca LEN=0012 Controller/PLCI/NCCI= 0x101 -- contr1#01: info element PI 81 88 contr1#01: In-band information available -- CAPI queue frame:[ TYPE: Control (4) SUBCLASS: Unknown control '14' (14) ] [contr1#01] INFO_IND ID=002 #0x08cb LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x18 InfoElement = 89 INFO_RESP ID=002 #0x08cb LEN=0012 Controller/PLCI/NCCI= 0x101 -- contr1#01: info element CHANNEL IDENTIFICATION 89 INFO_IND ID=002 #0x08cc LEN=0015 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x800d InfoElement = default INFO_RESP ID=002 #0x08cc LEN=0012 Controller/PLCI/NCCI= 0x101 -- contr1#01: info element SETUP ACK -- CAPI/contr1#01/103-8 is making progress passing it to CAPI/contr1#02/123-7 == contr1#02: Requested PROGRESS-Indication for CAPI/contr1#02/123-7 INFO_IND ID=002 #0x08ce LEN=0015 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x8002 InfoElement = default INFO_RESP ID=002 #0x08ce LEN=0012 Controller/PLCI/NCCI= 0x101 -- contr1#01: info element CALL PROCEEDING -- CAPI queue frame:[ TYPE: Control (4) SUBCLASS: Unknown control '15' (15) ] [contr1#01] -- CAPI/contr1#01/103-8 is proceeding passing it to CAPI/contr1#02/123-7 == contr1#02: Requested PROCEEDING-Indication for CAPI/contr1#02/123-7 INFO_IND ID=002 #0x08dd LEN=0015 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x8001 InfoElement = default INFO_RESP ID=002 #0x08dd LEN=0012 Controller/PLCI/NCCI= 0x101 -- contr1#01: info element ALERTING -- CAPI queue frame:[ TYPE: Control (4) SUBCLASS: Ringing (3) ] [contr1#01] -- CAPI/contr1#01/103-8 is ringing == contr1#02: Requested RINGING-Indication for CAPI/contr1#02/123-7 -- contr1#02: attempting ALERT in state 2 CAPI devicestate requested for contr1#01/103 INFO_IND ID=002 #0x0915 LEN=0020 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x29 InfoElement = 07 09 06 10 26 INFO_RESP ID=002 #0x0915 LEN=0012 Controller/PLCI/NCCI= 0x101 -- contr1#01: info element Date/Time 07/09/06 16:38 INFO_IND ID=002 #0x0916 LEN=0015 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x8007 InfoElement = default INFO_RESP ID=002 #0x0916 LEN=0012 Controller/PLCI/NCCI= 0x101 -- contr1#01: info element CONNECT CONNECT_ACTIVE_IND ID=002 #0x0918 LEN=0020 Controller/PLCI/NCCI= 0x101 ConnectedNumber = 00 83103 ConnectedSubaddress = default LLC = default CONNECT_ACTIVE_RESP ID=002
[asterisk-users] Different Networks
I have multiple upstreams in my office. The primary upstream is having some issues with latency\jitter. I want to move the VoIP traffic to another interface. I have the router set to send all traffic destined for local networks out the respective interfaces. Traffic destined to the Internet goes out one of the upstreams. I can do this on a per-IP basis and have successfully done so in testing on my laptop and a couple other machines. I also have it in production for an ATA. I also switch all devices to use another upstream with the failure of the primary ISP. Again, this works with everything but the Asterisk server. The internal Asterisk server cannot connect to the Asterisk server out on the public Internet. How do I investigate this? Here is the definition on the internal server: [rwestics] type=friend ;host=208.100.1.33 ;miho.ics-il.net host=dynamic ;username=rwestics secret=*** context=rwest disallow=all allow=ulaw Here is the definition on the public Internet server: [rwestics] type=friend host=dynamic ;username=ics secret=** qualify=yes context=outbound-scripted accountcode=12 callerid=* - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Users Conference Friday @ 12:30PM EDT
FRIDAY September 7th at 12:30 PM EDT http://www.asteriskusersconference.org for more information on how to listen, talk, or both :) This week, ENUM is the main subject, although our friends at e164.org haven't been able to talk to us as planned. Come on by and share what you know about ENUM or ask questions. Also, during Astricon, we are hoping people will call us with reports, either live or recorded and maybe someone will have some video? The IRC channel on Freenode.net is #asterisk_users_conference Past conference recordings: http://www.asteriskusersconference.org/topics.php rr ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium Innovation Awards
I justed wanted to send out a quick note and remind the Asterisk community about the Digium Innovation awards, as the dealine for submission is approaching. (Submissions are due by October 1st.) This is a great chance to show off your innovative Asterisk-based solutions, and to get recognized by Digium for your efforts. The entry form can be found at http://www.digium.com/en/company/awards/digium-innovation-application.doc -- Jared Smith Community Relations Manager Digium, Inc. Digium® Innovation Award The Innovation Award is designed to recognize developers, customers and partners for outstanding achievements that are improving business processes, overcoming technology challenges and enhancing the company’s bottom line. Benefits of being named the Digium Bold Innovation winner: * Two Complimentary passes for Digium|Asterisk® World * Participation in a VIP executive breakfast/dinner with Keynote speakers and honored guests * Presentation of the award with a profile of your company in the Conference General session * Chance to highlight the accomplishments of you and your team * Recognition by your industry, friends and family * Tour of Digium HQ In addition to the above, Digium Bold Innovation winners will receive: * Presentation at Digium|Asterisk World * Congratulatory press release from Digium, Inc. * Listing on the Digium website Award Categories: * Pioneer Award: Most innovative implementation * Big Biz Asterisk: Largest enterprise class installation * ROI: Best measurable ROI from implementing Asterisk based solution * Inside Out Award: Best use of Asterisk in a business outside of telecommunications Awards will be presented for each category on a global basis. Eligibility: * All Digium|Asterisk customers and partners world-wide with solutions that are live or in production * Submit as many projects as you like * Each project requires a separate submission Judges: * Asterisk Community member * Mark Spencer * Danny Windham * Bill Miller Selection Process: * What makes your Asterisk-based solution innovative? * Did your solution improve processes? * What technology challenges were overcome to achieve your goal? * What was the largest implementation of your solution? * What was the measurable ROI and competitive advantages generated by project? * What can be achieved with the solution today that couldn’t be previously accomplished? * Creativity * Presentation What are the submission dates? October 1 - Last day for submission October 15 – Category winners are announced Digium Innovation Award will be presented during a ceremony at Digium|Asterisk World in Boston, MA. October 30th 31st. Award applications must be submitted by October 1, 2007. Please send completed forms to Julie Webb via email at [EMAIL PROTECTED] or via postal service. Digium, Inc. ATTN: Julie Webb 150 West Park Loop Suite 100 Huntsville, AL 35806 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multitenant or Multiple virtual machines
Any opinions/comments/recommendations? Before anyone recommends just buying the virtual PBX service from someone else, we _really_ want to do this in-house :) I am all for using VPS (virtual private servers) instead of the classic multitenant. Here are some reasons: - a VPS provides a Linux environment to each client. This is a big plus for some clients knowing they are not boxed - Any custom development, new features can be easily applied to individual clients VPSes without destroying/affecting other clients. Remember, these clients must have their phone lines up in order to trade :) - Firmware updates, bug fixes failures etc, will not affect other tenants. - I can move clients VPS to another server, another data centre across the world if necessary with a couple commands. - One can allocate specific resources to each tenant. This is very important for call centres for example. All of the above, and a lot more commercial reasons have made me think of developing administration high availability solution for VPSes which we and our customers use extensively. Thanks Senad www.bicomsystems.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bridge on DIVA card and how to see it
On Thu, 6 Sep 2007, lemmel lemmel wrote: So please provide a log with set verbose 5 capi debug Well, let's go for the debug log :-). In this capture, I used only one adapter, and I performed a call to asterisk, which dialed a number (as described in the previous mail) ; so this is the log : -- Executing [EMAIL PROTECTED]:2] Dial(CAPI/contr1#02/123-7, CAPI/contr1/b:103||tT) in new stack ... -- CAPI queue frame:[ TYPE: Control (4) SUBCLASS: Answer (4) ] [contr1#01] -- CAPI/contr1#01/103-8 answered CAPI/contr1#02/123-7 == contr1#02: Requested Indication-STOP for CAPI/contr1#02/123-7 CAPI devicestate requested for contr1#01/103 Hmm, it looks like there is something missing. Just after the 'answered' message, asterisk should say something like 'Attempting native bridge'. But since it doesn't appear, I think for some reason asterisk itself doesn't want to bridge here. Is this the complete log? Maybe you want to provide a full log (including call of the first channel and the hangup) to my personal mail? Armin ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help needed - ISDN is redialling
We've just received a bill from bt where it claims that we are making numerous calls to the same number time after time. e.g. 01226xx Barnsley20/06/2007 211516:00:00 01226xx Barnsley20/06/2007 121908:55:32 01226xx Barnsley21/06/2007 211516:00:00 01226xx Barnsley21/06/2007 131508:00:00 01226xx Barnsley21/06/2007 051508:00:00 01226xx Barnsley22/06/2007 211516:00:00 01226xx Barnsley22/06/2007 051508:00:00 01226xx Barnsley22/06/2007 131508:00:00 01226xx Barnsley23/06/2007 211532:00:00 01226xx Barnsley23/06/2007 131508:00:00 01226xx Barnsley23/06/2007 051508:00:00 01226xx Barnsley24/06/2007 211516:00:00 01226xx Barnsley24/06/2007 051508:00:00 01226xx Barnsley24/06/2007 131508:00:00 01226xx Barnsley25/06/2007 211516:00:00 01226xx Barnsley25/06/2007 051508:00:00 01226xx Barnsley25/06/2007 131508:00:00 01226xx Barnsley26/06/2007 211516:00:00 01226xx Barnsley28/06/2007 211516:00:00 01226xx Barnsley28/06/2007 131508:00:00 01226xx Barnsley28/06/2007 051508:00:00 01226xx Barnsley29/06/2007 211532:00:00 01226xx Barnsley29/06/2007 131508:00:00 01226xx Barnsley29/06/2007 051508:00:00 01226xx Barnsley30/06/2007 211516:00:00 01226xx Barnsley30/06/2007 051508:00:00 01226xx Barnsley30/06/2007 131508:00:00 01226xx Barnsley01/07/2007 211510:22:42 All our calls are made using the asterisk AMI, and as far as our records are concerned, the original call was hung up after 5 minutes. Notice that most of these calls are 8, 16 or 32 hours long ! We are using a EuroISDN and a sangoma A102 pri card, with asterisk 1.4 svn trunk. Has anyone seen anything like this, or could it be a BT fault ? Julian. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bridge on DIVA card and how to see it
Maybe you want to provide a full log (including call of the first channel and the hangup) to my personal mail? I just added an attachment to this mail. Thanks a lot :-) P.S.: this log was generated with verbosity to 5 and with capi debug _ Découvrez le Blog heroic Fantaisy d'Eragon! http://eragon-heroic-fantasy.spaces.live.com/ *CLI CONNECT_IND ID=002 #0x168a LEN=0041 Controller/PLCI/NCCI= 0x401 CIPValue= 0x10 CalledPartyNumber = 81123 CallingPartyNumber = 00 83107 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = 91 81 AdditionalInfo BChannelinformation= default Keypadfacility = default Useruserdata = default Facilitydataarray = default SendingComplete= default -- CONNECT_IND (PLCI=0x401,DID=123,CID=107,CIP=0x10,CONTROLLER=0x1) contr1#02: msn='*' DNID='123' DID == contr1#02: setting format alaw - 0x8 (alaw) == contr1#02: Incoming call '107' - '123' INFO_IND ID=002 #0x168b LEN=0019 Controller/PLCI/NCCI= 0x401 InfoNumber = 0x70 InfoElement = 81123 *CLI INFO_IND ID=002 #0x3530 LEN=0036 Controller/PLCI/NCCI= 0x301 InfoNumber = 0x1c InfoElement = 91 a1 12 02 01t02 01 2200a a1 05003 02 01 00 82 01 01 INFO_RESP ID=002 #0x3530 LEN=0012 Controller/PLCI/NCCI= 0x301 -- contr1#01: info element FACILITY INFO_IND ID=002 #0x3531 LEN=0017 Controller/PLCI/NCCI= 0x301 InfoNumber = 0x1e InfoElement = 81 88 INFO_RESP ID=002 #0x3531 LEN=0012 Controller/PLCI/NCCI= 0x301 -- contr1#01: info element PI 81 88 contr1#01: In-band information available INFO_IND ID=002 #0x3532 LEN=0017 Controller/PLCI/NCCI= 0x301 InfoNumber = 0x8 InfoElement = 80 90 INFO_RESP ID=002 #0x3532 LEN=0012 Controller/PLCI/NCCI= 0x301 -- contr1#01: info element CAUSE 80 90 INFO_IND ID=002 #0x3533 LEN=0015 Controller/PLCI/NCCI= 0x301 InfoNumber = 0x8045 InfoElement = default INFO_RESP ID=002 #0x3533 LEN=0012 Controller/PLCI/NCCI= 0x301 -- contr1#01: info element DISCONNECT -- contr1#01: Disconnect case 1 -- CAPI queue frame:[ TYPE: Control (4) SUBCLASS: Hangup (1) ] [contr1#01] == contr1#01: CAPI Hangingup for PLCI=0x301 in state 2 -- contr1#01: activehangingup (cause=16) for PLCI=0x301 DISCONNECT_B3_REQ ID=002 #0x31d3 LEN=0013 Controller/PLCI/NCCI= 0x260301 NCPI= default == Spawn extension (standardtelephonique, 103, 2) exited non-zero on 'CAPI/contr1#02/123-10' == contr1#02: CAPI Hangingup for PLCI=0x401 in state 2 -- contr1#02: activehangingup (cause=16) for PLCI=0x401 DISCONNECT_B3_REQ ID=002 #0x31d4 LEN=0013 Controller/PLCI/NCCI= 0x250401 NCPI= default CAPI devicestate requested for contr1#01/103 CAPI devicestate requested for contr1#01/103 CAPI devicestate requested for contr1#02/123 CAPI devicestate requested for contr1#02/123 DISCONNECT_B3_CONF ID=002 #0x31d3 LEN=0014 Controller/PLCI/NCCI= 0x260301 Info= 0x0 DISCONNECT_B3_CONF ID=002 #0x31d4 LEN=0014 Controller/PLCI/NCCI= 0x250401 Info= 0x0 DISCONNECT_B3_IND ID=002 #0x3535 LEN=0015 Controller/PLCI/NCCI= 0x260301 Reason_B3 = 0x0 NCPI= default DISCONNECT_B3_RESP ID=002 #0x3535 LEN=0012 Controller/PLCI/NCCI= 0x260301 DISCONNECT_REQ ID=002 #0x31d5 LEN=0013 Controller/PLCI/NCCI= 0x301 AdditionalInfo = default DISCONNECT_B3_IND ID=002 #0x3536 LEN=0015 Controller/PLCI/NCCI= 0x250401 Reason_B3 = 0x0 NCPI= default DISCONNECT_B3_RESP ID=002 #0x3536 LEN=0012 Controller/PLCI/NCCI= 0x250401 DISCONNECT_REQ ID=002 #0x31d6 LEN=0013 Controller/PLCI/NCCI= 0x401 AdditionalInfo = default DISCONNECT_CONF ID=002 #0x31d5 LEN=0014 Controller/PLCI/NCCI= 0x301 Info= 0x0 INFO_IND ID=002 #0x3537 LEN=0015 Controller/PLCI/NCCI= 0x301 InfoNumber = 0x805a InfoElement = default INFO_RESP ID=002 #0x3537 LEN=0012 Controller/PLCI/NCCI= 0x301 -- contr1#01: info element RELEASE COMPLETE DISCONNECT_IND ID=002 #0x3539 LEN=0014 Controller/PLCI/NCCI
[asterisk-users] Skype + Asterisk
Has anybody ever integrated Skype with Asterisk? If you have, which software would you recommend to accomplish such a task? ChanSkype? And how reliable are the calls? Did the DTMF tones work? Thanks in advance. _ Discover sweet stuff waiting for you at the Messenger Cafe. Claim your treat today! http://www.cafemessenger.com/info/info_sweetstuff.html?ocid=TXT_TAGHM_SeptHMtagline2 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Dial tone came from fxs modules
Mojo with Horan Company, LLC wrote: Just to be clear, I thought that dialtone provision didn't require the power cable, just generating ring voltages? Can anyone say? The DC-DC converter on the FXS modules supplies both ringing voltage and line voltage. If the power connector is not plugged into the TDM card then the FXS module can't generate line current and the call will not be held up. (From Mickey Morris, hardware design engineer here at Digium) Matthew Fredrickson Moj Anthony Messina wrote: On Wednesday 05 September 2007 09:09:25 am wassim darwish wrote: Hi: I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when i made modprobe wctdm the fxs modules is lightened but there is no dial tone came from it . Can i get some help please. do you have the power cable attached to it. that's what you need to generate a dialtone. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bridge on DIVA card and how to see it
On Thu, 6 Sep 2007, lemmel lemmel wrote: Maybe you want to provide a full log (including call of the first channel and the hangup) to my personal mail? I just added an attachment to this mail. Thanks a lot :-) P.S.: this log was generated with verbosity to 5 and with capi debug Your Dial string has errors: CAPI/contr1/b:103||tT b: sets the caller number to 'b'. I think what you are trying to do is CAPI/contr1/103/b I'm not sure, but maybe tT lets asterisk avoiding the bridge. Armin ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multitenant or Multiple virtual machines
On Thu, Sep 06, 2007 at 04:38:40PM +0100, Senad Jordanovic wrote: Any opinions/comments/recommendations? Before anyone recommends just buying the virtual PBX service from someone else, we _really_ want to do this in-house :) I am all for using VPS (virtual private servers) instead of the classic multitenant. Here are some reasons: - a VPS provides a Linux environment to each client. This is a big plus for some clients knowing they are not boxed - Any custom development, new features can be easily applied to individual clients VPSes without destroying/affecting other clients. Remember, these clients must have their phone lines up in order to trade :) - Firmware updates, bug fixes failures etc, will not affect other tenants. But have to be tested and applied separately to each one = more work. - I can move clients VPS to another server, another data centre across the world if necessary with a couple commands. - One can allocate specific resources to each tenant. This is very important for call centres for example. Allocating resources means that the global pool, which is normally not used, can't easily be shared. This can be a pain. Deviding the memory of a 2GB server between 8 tenants gives you 8 258MB servers. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Problem with International Calls
Hello All, Does anyone knows a good carrier who can pass DTMF tone while doing Call Back? Currently, the Call Back system works within US, but as soon as international users tries to enter phone number the system does not understand the tones. I tried to change the sip config to inband, auto, RFC2833 but it didnt work... So I suspect its my VoIP Carrier who doesn't pass the International DTMF tones. Any suggestions? Cheers, Nitesh ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dead SIP channels
I am using a2billing as calling card platform with asterisk 1.2.17. After running for several days, if I issue 'sip show channels' command, I got a lot of dead sip channels although 'show channels' command only show 5 channels. What cause these dead channels? How can I clean out these dead channels? Will they pose any problem to my * server if left alone? What does this (d) mean? Here is the output from 'sip show channels': Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 195.7.123.234 +180924402 3c3c4cee419 00102/0 alaw No Tx: ACK 9.9.94.9 6478517573 2752611-195 00101/1 ulaw No Rx: ACK 136.59.30.19 8787041796 76775e35788 00102/0 ulaw No Tx: ACK 9.9.95.13 9057047798 2752419-199 00101/1 ulaw No Rx: ACK 195.7.123.234 +011503733 25afde8070b 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011503733 71688696061 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011503733 1700ab8b2ae 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011578435 0ecb33f75bb 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011962642 71eac20715c 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011962642 01b9eacf6de 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011962642 744e7a3f501 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011962642 0080443e6ad 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011962642 6f3745a266d 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011221693 3b705a03141 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011595981 4ab469132b7 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011595981 0b2dcf2332b 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011595981 583bd73d09a 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011593222 4d237ba325e 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011639103 33f84238290 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011526778 72bd7b5f080 00102/2 unkn No (d) Rx: BYE 195.7.123.234 +011527693 0ffa93c642d 00102/2 unkn No (d) Rx: BYE gary___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multitenant or Multiple virtual machines
Tzafrir Cohen wrote: On Thu, Sep 06, 2007 at 04:38:40PM +0100, Senad Jordanovic wrote: Any opinions/comments/recommendations? Before anyone recommends just buying the virtual PBX service from someone else, we _really_ want to do this in-house :) I am all for using VPS (virtual private servers) instead of the classic multitenant. Here are some reasons: - a VPS provides a Linux environment to each client. This is a big plus for some clients knowing they are not boxed - Any custom development, new features can be easily applied to individual clients VPSes without destroying/affecting other clients. Remember, these clients must have their phone lines up in order to trade :) - Firmware updates, bug fixes failures etc, will not affect other tenants. But have to be tested and applied separately to each one = more work. Since this is custom development for client... Client is happy to pay for it... Next.. - I can move clients VPS to another server, another data centre across the world if necessary with a couple commands. - One can allocate specific resources to each tenant. This is very important for call centres for example. Allocating resources means that the global pool, which is normally not used, can't easily be shared. This can be a pain. Deviding the memory of a 2GB server between 8 tenants gives you 8 258MB servers. True, but done corectly it is huge benefit/saving. Just the fact that virtual machines/VPS technologies is now supported in kernels of many operating systems tells A LOT... Senad ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multitenant or Multiple virtual machines
A question. are the clients going to be able to manage the PBX? or are you going to give them the PBX service without access to each server? On 9/6/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Sep 06, 2007 at 04:38:40PM +0100, Senad Jordanovic wrote: Any opinions/comments/recommendations? Before anyone recommends just buying the virtual PBX service from someone else, we _really_ want to do this in-house :) I am all for using VPS (virtual private servers) instead of the classic multitenant. Here are some reasons: - a VPS provides a Linux environment to each client. This is a big plus for some clients knowing they are not boxed - Any custom development, new features can be easily applied to individual clients VPSes without destroying/affecting other clients. Remember, these clients must have their phone lines up in order to trade :) - Firmware updates, bug fixes failures etc, will not affect other tenants. But have to be tested and applied separately to each one = more work. - I can move clients VPS to another server, another data centre across the world if necessary with a couple commands. - One can allocate specific resources to each tenant. This is very important for call centres for example. Allocating resources means that the global pool, which is normally not used, can't easily be shared. This can be a pain. Deviding the memory of a 2GB server between 8 tenants gives you 8 258MB servers. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Register Extension
Anybody uses Asterisk Java to register an extension to an Asterisk server ??? Is there any solution for this ? Phan Anh Vu DT12.K49.HUT RDLab ( C9.410 ) HUT - Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out.___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multitenant or Multiple virtual machines
Edgar Guadamuz wrote: A question. are the clients going to be able to manage the PBX? Yes... or are you going to give them the PBX service without access to each server? Up to you... Senad ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bridge on DIVA card and how to see it
Your Dial string has errors: CAPI/contr1/b:103||tT b: sets the caller number to 'b'. I think what you are trying to do is CAPI/contr1/103/b I just checked the README, and I saw this, thanks :-) (the docs I readed are a bit old I suppose -e.g. http://www.voip-info.org/wiki/view/Asterisk+CAPI+readme- I'll correct them later) I'm not sure, but maybe tT lets asterisk avoiding the bridge. I'll test tomorrow (I leaved office :-), I am GMT+1). ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] off-hook warning tone
Eric ManxPower Wieling wrote: The correct term for this tone is howler. I'm surprised it is not in indications.conf I recall seeing it there once, but I'm reaching into the dusty recesses of my memory right now. I noticed that all the replies to the OP assumed a SIP handset. The howler only applies to analog sets. I've made the same observation -- Asterisk is supposed to send a howler, but my phones just a get a wimpy fast busy when left off hook. Once one of our analog sets was left off hook for nearly a day before anybody noticed (How come we're not getting any calls?) How do I make this work the way it's supposed to? -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cascading queues calls not joining unavailable queues.
Hi! - Trying a repost, my first message didn't seem to make the list. I have one main queue with agents that take calls to our main phonenumber. Now i want to cascade calls through to the fallback queue immediately when all the agents in the first queue are 'unreachable' in any way (be it OffHook, DND, Paused, etc...) Somehow calls still keep hanging around in the main queue even if agents are Busy or 'DND' for the specified timeout before returning to the dialplan which then calls the next queue. The extensions.conf section that places the call on the main queue and afterwards the second queue: | exten = 511,n,Queue(511,t,,,30) ; Main queue | exten = 511,n,Queue(611,t,,,30) ; Fallback queue And this is my queues.conf accordingly. I'll only show the main queue config, as the fallback queue config is EXACTLY the same, except for the queuemembers ofcourse. | [511] | servicelevel = 30 | announce = voice/connected | musiconhold = default | strategy = ringall | context = vanuit-queues | timeout = 10 | wrapuptime = 10 | announce-frequency = 10 | announce-holdtime = no | joinempty = strict | leavewhenempty = yes | member = SCCP/206 | member = SCCP/210 This selection of loglines shows Asterisk is aware that noone is answering the queue: | logger.c: -- Goto (groepen,511,1) | logger.c: -- Called SCCP/210 | logger.c: -- Called SCCP/206 | logger.c: -- SCCP/206 is busy | logger.c: -- SCCP/210 is busy | app_queue.c: No one is answering queue '511' (7/2/0) | logger.c: -- Stopped music on hold on SIP/10.10.1.1 | logger.c: -- Told SIP/10.10.1.1 in 511 their queue position (which was 1) | logger.c: -- Started music on hold, class 'default', on SIP/10.10.1.1 | logger.c: -- Called SCCP/210 | logger.c: -- Called SCCP/206 | logger.c: -- SCCP/206-0021 is busy | logger.c: -- SCCP/210-0020 is busy | app_queue.c: No one is answering queue '511' (7/2/0) | [ ... etc ... ] What am i doing wrong here? Can anyone shed some light? Thanks! Sander. -- | The short fortune teller who escaped from prison: a small medium at large. | 1024D/08CEC94D - 34B3 3314 B146 E13C 70C8 9BDB D463 7E41 08CE C94D -- | The story of my life; warm beer and cold women. | 1024D/08CEC94D - 34B3 3314 B146 E13C 70C8 9BDB D463 7E41 08CE C94D ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Build your own appliance concept
I've been working on this the past few days and thought I would put it out there to see if anyone else has interest in it. It really has nothing to do with the Digium appliance, I've just been looking for some mass produced solid state hardware to run small branch offices off of for awhile now and I think I've finally landed on something I like. Basically I've taken an HP thin client workstation which is all solid state and loaded Debian and Asterisk on it (well, Asterisk-GUI too, but just to prove I could make it appliance-worthy). I'd be interested in any feedback on how to improve it, specifically on how to make Debian and Asterisk take up less space so I could buy the model that only has 512 MB of flash rather than 1 GB. Here's the link. http://tinyurl.com/2hf2cu Let me know what you think. Jeremy ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inbound SIP issues
I have an issue with receiving inbound calls. I've got bandwidth.com trunks incoming to my asterisk box, bandwidth sends all incoming traffic to one of two IP addresses, and requires outbound traffic go to either of the same two IP addresses. I've got to use fromuser=DID on outgoing calls so they apply the right caller ID. My issue is that I want incoming calls to match on a specific sip.conf entry, but they are matching on my outgoing entries and dropping(I don't have context associated with them). Here's relevant sip.conf entries -- [bandwidth_inbound_1] host=4.79.212.236 port=5060 type=peer disallow=all allow=ulaw dtmfmode=rfc2833 canreinvite=no nat=no context=frombandwidth [bandwidth_inbound_2] host=216.82.224.202 port=5060 type=peer disallow=all allow=ulaw dtmfmode=rfc2833 canreinvite=no nat=no context=frombandwidth [bandwidth_outbound_did1] host=4.79.212.236 port=5060 type=peer disallow=all allow=ulaw dtmfmode=rfc2833 canreinvite=no reinvite=no nat=no fromuser=did1 If calls come in from 4.79.212.236 they are immediately matched to context [bandwidth_outbound_did1] If I put the inbound contexts under the outbound in sip.conf they work, is that the design intention of sip.conf? Bandwidth doesn't require or accept register statements, so I can't use that to send calls to specific extensions. Is there any easier logic to attach my fromuser when I have multiple DIDs? Ideally I'd love 2 entries for them total. I'm running asterisk 1.4.11 if it helps. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Testing Framework
I think the testing frame work includes both the components and system testing. I wish to add some more test even though all giants may aware, since i wish to do some contribution to asterisk what ever i can. i am plannig for the framework and addon as given below, expecting techies advise in this, Testing frame work may contain testing internally and externally, i mean to say internally, some client may stick into the problem of voice quality, when more phones on the PBX, that time, we may not leave the system from the network, their call may get interrupted, that time internal testing daemon will work for the performance analysis till the lelvel of without affecting present calls. externally means PBX system on production for performance analysis, as per the test case started in the mail. internal and external testing may be configurable at the run time, through some verbose like variable configurable at run time, addon should have the capability of releasing the testing call bandwidth, whenever PBX gets new call, this might be simple example. procesding with Testing frame work requirements, call on volume, 1. SIP - SIP, multiple SIP clients support, through which we can either direct the call for testing to another client registered in testing frame work, or return back to the different client registered in the same testing frame work, when the call incoming and outgoing call are handled in single framework point, wave analysis also cane be done with the script and performance can be easily evaluated. On production performance testing, by connecting multiple testing framework point to the PBX, having the sending files in all the frame work, analysis and performance evaluation can be done very easily, i also think that once it is done for SIP with compatiblity of like channel driver, we can adapt IAX2, anything we want. i think this type of testing would make the system stable and provide good support on system on running also. Hariharan.V. RD Engineer, NEEVEE Technologies, On 9/3/07, dave cantera [EMAIL PROTECTED] wrote: matt, are you looking for unit testing of the * components or systems testing, testing the finished product? or both? I think you are onto something here... I hope it takes root. I would say put it in the addons. it would be Great if digium takes it up. it is a smart move for them to foster, cajole, nudge, and support it. call volume I would leave to others as different processors, O/S, builds, kernel versions, and configurations will have too many variables. I was playing with the idea of monitoring multiple * systems. perhaps we can start out with testing the components and then migrate the project (future) to one pbx monitor the other. we will need scripts to initiate some action, config to make some measurements, the scripts to gather the results into a nice neat little summary report. you will want to take the human aspect out of the picture as much as possible. for example: on pbx A * create a recording in multiple formats .gsm, .wav, etc. * initiate a script to generate 5,10, or 25 calls to pbx B and play the file on pbx B * pbx B gets the calls, records them, * copy the recordings from pbx A to pbx B (or have that already done) * have a wave analyzer compare the recordings to the original files (you know I won't be writing that program! :) * report on anomalies *call * *Technology * *recording delta * 1 Zap Provider 1 2% 2 VoIP Provider 2 5% 3 VoIP Provider 2 15% ... VoIP Provider 3 ... let me know what you think! daveC Matt Riddell wrote: Hash: SHA1 Hi, So, now that we've all complained about the state of testing of Open Source versions of Asterisk, lets do something about it. I propose we start with a list of things that we think should be tested in Asterisk, and means to test them. Maybe we could run certain tests based on the changes between minor versions? Anyway lets start. Call Volumes 1) Call volume up to x channels from SIP to SIP (i.e. sipp) 2) Call volume up to x channels from IAX2 to SIP 3) Call volume up to x channels from IAX2 to IAX2 Application testing 4) Connect x calls between techs to Meetme (leave running for 1 hour) 5) Connect x concurrent calls to VoiceMail Call Centre Testing 6) Send x calls to a queue with no agents in it, leave them holding for x minutes 7) Run x calls against AMD connected to recorded known good files Recording 8) Run x calls recording simultaneously from an automatically generated call, play ulaw/alaw - compare outputs. You get the idea. If people can add to this list, I can start making a few scripts and programs that will test them (as I'm sure others can). If we end up with a complete list, I'm sure some of our
Re: [asterisk-users] Build your own appliance concept
Jeremy P wrote: Basically I've taken an HP thin client workstation which is all solid state and loaded Debian and Asterisk on it (well, Asterisk-GUI too, but just to prove I could make it appliance-worthy). I'd be interested in any feedback on how to improve it, specifically on how to make Debian and Asterisk take up less space so I could buy the model that only has 512 MB of flash rather than 1 GB. I did something similar with a HP T5500, but I pulled out the flash memory and replaced it with a laptop hard drive. The connector was nothing more then a standard IDE connector. HP sells an expansion chassis for the unit that allows for a PCI card to be installed. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Udev issue on zaptel install
Debian GNU/Linux 3.1 (Sarge). This version supports udev 0.056-3 , but it is not installed as a normal part of the setup process. Which is my problem...probably. Now I have to figure how to set this up. Craig smime.p7s Description: S/MIME cryptographic signature ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Random Double Digits
We have a Asterisk box acting as a voicemail system and greeting/ call director for our phone system (NEC system). The problem we are having is that randomly (though most especially with cell phones) asterisk thinks it is getting a double digit. For example, somebody will enter 269 and asterisk will read 2269. I believe the core problem is the NEC system's volume as we've had problem with volumes for over a year, but short term I would like to find a solution while we try to solve the cause. Is there a way to tweak the zaptel settings so that Asterisk (or zaptel or whatever) better handles our situation? I realize there is probably not a single switch to turn on and I might have to do trial and error stuff, but we are willing to spend some time tweaking to find the best solution. My theory (though hard to prove since it is a random problem) is that the doubled-digit is being detected twice due to a drop out in either volume or (in the case of a cell-phone) poor connection. I have to this point never seen unexpected digits (somebody dials 269 and it read 249 or something like that), but we have seen both double-digits and missing digits (they dial 269 and it only reads one of the digits). Daniel ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Build your own appliance concept
On 9/6/07, Jeremy P [EMAIL PROTECTED] wrote: I've been working on this the past few days and thought I would put it out there to see if anyone else has interest in it. It really has nothing to do with the Digium appliance, I've just been looking for some mass produced solid state hardware to run small branch offices off of for awhile now and I think I've finally landed on something I like. Basically I've taken an HP thin client workstation which is all solid state and loaded Debian and Asterisk on it (well, Asterisk-GUI too, but just to prove I could make it appliance-worthy). I'd be interested in any feedback on how to improve it, specifically on how to make Debian and Asterisk take up less space so I could buy the model that only has 512 MB of flash rather than 1 GB. Here's the link. http://tinyurl.com/2hf2cu Let me know what you think. Jeremy Jeremy, AstLinux has no problem fitting in 512mb of flash: http://www.astlinux.org I've got 1.4 with Asterisk GUI in a development branch. Still around 30MB! -- Kristian Kielhofner ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Udev issue on zaptel install
On Thu, Sep 06, 2007 at 12:48:44PM -0700, Markham, Craig (FRTC Contractor) wrote: Debian GNU/Linux 3.1 (Sarge). This version supports udev 0.056-3 , but it is not installed as a normal part of the setup process. Which is my problem...probably. Now I have to figure how to set this up. udev is not a prerequirement for zaptel. Debian Sarge uses devfs by default, and Zaptel supports devfs as well. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cascading queues calls not joining unavailable queues.
| app_queue.c: No one is answering queue '511' (7/2/0) The 7/2/0 indicates that you have 7 members in your queue and 2 are busy. This would indicate that even though those 2 members are busy, there are still 5 more available members for taking calls. Since there are available members, you stay in the queue. Have you added additional queue members besides the ones you specified in queues.conf? Mark Michelson ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Build your own appliance concept
On Thu, Sep 06, 2007 at 01:05:28PM -0600, Jeremy P wrote: I've been working on this the past few days and thought I would put it out there to see if anyone else has interest in it. It really has nothing to do with the Digium appliance, I've just been looking for some mass produced solid state hardware to run small branch offices off of for awhile now and I think I've finally landed on something I like. Basically I've taken an HP thin client workstation which is all solid state and loaded Debian and Asterisk on it (well, Asterisk-GUI too, but just to prove I could make it appliance-worthy). I'd be interested in any feedback on how to improve it, specifically on how to make Debian and Asterisk take up less space so I could buy the model that only has 512 MB of flash rather than 1 GB. Here's the link. http://tinyurl.com/2hf2cu Let me know what you think. If you're going to build an appliance which is limited with disk space, then building Asterisk on it is generally not the best idea. You should have a separate build system. To create a separate Debian system under any other Linux system, use debootstrap. This will give you a build environment for a Debian system. I would generally recommend to use packages for as much as possible of the build process, as this allows a more reproducable build. Astlinux was also mentioned. If you decide to go that route, you'll probably need a separate build system as well, as you'll need to customize their image. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MINNESOTA: TwinCities Asterisk Users Group Meeting - This Saturday Sep 8th, 2007 (Only hours away)
To: Twin Cities Asterisk Users From: [EMAIL PROTECTED] Subject: TwinCities Asterisk Users Group Meeting this Saturday - Only 1 and 1/2 days away! Meeting Start: 09/08/2007 - 11:30am Hello all Twin Cities Asterisk Users, It's time once again to have another meeting. I've not had much time to prepare, but I'd really like to review and install with the group at our next meeting the software package formerly known as OpenPBX and now known as CallWeaver. This release now claims to have full t.38 support for faxing over IP. This has got to be one of the biggest issue facing professional installations at many business locations. OpenPBX and now CallWeaver are software forks of Asterisk. Lets think of the meeting as visiting your best friends ex! (Boy, is s/he looking hot?!) Lets review without getting caughtgrins... Basically this next meeting is a build fest. Please bring your projects and questions with you and as a group, we'll get you up and running. If you're interested in Faxing support, installing voip as your profession [or part of], or need Asterisk at your business, this is one meeting you do not want to miss. I'm going to attempt a CallWeaver installation, Live, with no editing or rehearsal. Be sure to get here early as the chairs go fast. The September 8th meeting will be starting at 11:30am. 7839 12th Ave S. Bloomington MN 55425 To get a head start on what you'll see, please visit: http://callweaver.org/blog I also want to say a special thanks to those who make our meetings possible and donate time, products, materials, give aways and so forth. Specifically Polycom for the unforgettable gifts, Cylogistics and Octasic for the Echo Cancel software licenses from last month, O'reilly for the gifts and all their support for user groups of all kinds and the special offers for members of our TCAUG group. (Ask me about these at a meeting sometime). All of our members who attend meetings too! Without you, what the heck am I doing here! To Everyone, a very special THANK YOU! Eric Osterberg, Sound Choice Communications LLC (651)-999-0888 - Voice Line PS: * Asterisk is a registered trademark of Digium. With great respect we use the name. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting Asterisk to Alcatel OmniPCX
Greetings list, I've been asked by someone to help them set up a SIP link between an asterisk system and an Alcatel OmniPCX (v6 software). The asterisk bit's fine, but I know nothing about the Alcatel except that it does apparently allow the setup of SIP trunks. Does anyone have experience with the Alcatel unit? Any obvious pitfalls to watch out for? Any suggestions gratefully appreciated. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to DUNDi branch office with area code?
hi: i am new to asterisk and dundi. we have some branch office which will use asterisk in the future. they will form a full-mesh structure so every site can contact each other directly. i want to try setup dundi, then we don't need to modify every pbx when a new site add in the cloud. thanks to the great dundi document caveman can do it and other resource in the voip-info.org. i learn the basic setup of dundi. but i want to a little advanced setup with area code. like this: site HQ: has extension 101,102,103, and site HQ has area code 99 site A: has extension 101,102,103, and site A has area code 01 site B: has extension 101,102,103 and site B has area code 02 site C: has extension 101,102,103 and site C has area code 03 we want to use 4 as prefix to call to the internal cloud. so user at site A can call 4-99-101 to contact extension 101 at HQ. site B can call 4-03-102 to contact extension 102 at site C. now i m confused about this structure with DUNDi. i don't know the best way to setup DUNDi for this structure. i think maybe i should do below when user call 4-99-101 at site A : 1. site A ask for dundi request 4-99-101 to site HQ 2. site HQ strip 4-99 and look up 101 at local context 3. site HQ return the destination to site A 4. site A use the destination to call extension 101 at site HQ i don't know if step 23 is possible in dundi.conf. the example in the internet didn't tell how to do this. or there are better/standard ways to do this? thanks a lot for any suggestion!! Regards, tbskyd ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral's Allison is having troublespeaking clearly
Kai-Uwe Jensen wrote: How are you playing the voice? Do you use something like app_swift or app_cepstral? Just fixed app_swift for my own installation by changing the framesize constant definition from 160*4 to 20, after googling for a similar issue. Works like a charm now. It only broke recently, i.e. not with the first 1.4.x releases, but maybe only a couple of months ago. Can you specify exactly where you made this change? I'm looking at the source for app_swift-0.9 right now and don't see a framesize constant. I'm getting some breakup when using app_swift over an IAX connection and thought I'd try this. Thanks Steve ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral's Allison is having troublespeaking clearly
Sure. Sorry to be unclear about it. I was using app_swift-2.0rc1, from http://www.mezzo.net/asterisk/app_swift.html. Part of that package is app_swift.c. At line 68, I changed the declaration const int framesize = 160*4; to const int framesize = 20; That fixed things here. As it seems, that fix may not work (or even be appropriate) for app_swift-0.9, which I would assume you got from loopfree.net. On 9/6/07, Steve Prior [EMAIL PROTECTED] wrote: Can you specify exactly where you made this change? I'm looking at the source for app_swift-0.9 right now and don't see a framesize constant. I'm getting some breakup when using app_swift over an IAX connection and thought I'd try this. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Build your own appliance concept
On Thu, 6 Sep 2007, Jeremy P wrote: I've been working on this the past few days and thought I would put it out there to see if anyone else has interest in it. It really has nothing to do with the Digium appliance, I've just been looking for some mass produced solid state hardware to run small branch offices off of for awhile now and I think I've finally landed on something I like. Basically I've taken an HP thin client workstation which is all solid state and loaded Debian and Asterisk on it (well, Asterisk-GUI too, but just to prove I could make it appliance-worthy). I'd be interested in any feedback on how to improve it, specifically on how to make Debian and Asterisk take up less space so I could buy the model that only has 512 MB of flash rather than 1 GB. I built my own appliance some time back - initially for a router project, but I've since adapted it for Asterisk boxes and NAS boxes.. The basic unit has 64MB of IDE-flash, 256MB (or more) RAM. The flash IDE device has one partition and is bootable, so it has a /boot with a bzImage in it, enough of a /dev/ and /etc to make Lilo work on it and an initrd.gz which is unpacked into a 128MB RAM disk, then the system runs entirely from RAM once booted, so there's no continual write to flash issues (I hope!) I do actually have a 2nd partition on the device which I tar all the configuration files into - the bare minimal of what I need gets stored there whenever something changes. (and a copy of astdb too). I don't think this is perfect, and is prone to issues like a power cycle during write, but ... I put a 2nd IDE flash device for Voicemail storage - that does have a live filesystem on it (currently just ext2, which I force an fsck of at boot time, if it's dirty) I've used 64MB to 256MB devices for this (storing VM in GSM format only), some customers want call recording, so they get the bigger ones, but I'm thinking of moving to a laptop drive for people who want even more (and enable idle spin down, etc.) I build the kernel and initrd.gz file on a separate box - it's Debian, but it could be anything as I don't actually put a distribution as such into it, I just copy the files I need, and I'm lazy about it, so I copy all of /bin, /lib, most of /etc and a /dev and selected bits of /usr/bin and /usr/lib. (I use ldd on all the executables to work out which libraries I really need from /usr/lib) The kernel is a custom kernel for the hardware with no modules apart from Zaptel, etc. I copy everything into a 128MB file, zeroed (it compresses better) formatted ext2, mounted as a lookback device. Once the copy is complete, I unmount it, gzip -9 it and that's the initrd.gz file. You need to make sure that the Linix kernel you compile has the ability to load an initrd.gz file and a big enough ramdisk! It's not that efficient, and I could save space by using uClib, busybox, etc. but it's really not worth it, but 2 things I don't have on the target system is perl and vim.. Perl is about 10MB, as is vim. Right now I don't have a need for either (and I use nano when I do need to tweak stuff which is rarely) Perl would be nice so I could run stuff like mrtg locally on the boxes, but isn't essential for now. So if there are some new security implications on the current Debian, or an asterisk upgrade, I just upgrade/update the build box, then create a new initrd.gz file and install it. (however this is in the order of 40MB for an Asterisk system with apache php) so it a bit tricky to do a field upgrade if the remote system is bandwidth limited, but I can pull it in off a USB drive if necessary. My /etc/asterisk and /var/www/docs are actually stored as part of the tar file, so upgrading those is fairly trivial. This is what a running system looks like: $ df -h FilesystemSize Used Avail Use% Mounted on /dev/ram0 124M 107M 18M 87% / tmpfs 125M 0 125M 0% /dev/shm /dev/hdc2 60M 23M 37M 39% /data If I mount the flash device, then: # ls -l /mnt total 39019 drwxr-xr-x 2 root root 1024 Aug 9 14:54 boot drwxr-xr-x 13 root root24576 Dec 6 2006 dev drwxr-xr-x 2 root root 1024 Nov 15 2006 etc -rw-r--r-- 1 dsx 1000 39758472 Aug 9 14:53 image.gz drwx-- 2 root root12288 Dec 12 2006 lost+found # ls -l /mnt/boot total 2849 -rw-r--r-- 1 root root 512 Dec 12 2006 boot.0300 -rw-r--r-- 1 root root 512 Dec 22 2006 boot.0800 -rw-r--r-- 1 root root 512 Dec 12 2006 boot.1600 -rw-r--r-- 1 dsx 1000 1390066 Jun 5 15:47 bzImage -rw--- 1 root root 31744 Aug 9 14:54 map -rw-r--r-- 1 root root 98728 Sep 21 2006 memtest86+.bin -rw-r--r-- 1 root root 241 Oct 28 2006 message Because everything is in RAM, it's actually quite fun to play with trying to destroy it :) Eg. # cd / ; rm -rf * then just reboot it to recover... The one thing that's not appliance about it is the box - it's still a PC at
Re: [asterisk-users] Build your own appliance concept
I've been working on this the past few days and thought I would put it out there to see if anyone else has interest in it. It really has nothing to do with the Digium appliance, I've just been looking for some mass produced solid state hardware to run small branch offices off of for awhile now and I think I've finally landed on something I like. Basically I've taken an HP thin client workstation which is all solid state and loaded Debian and Asterisk on it (well, Asterisk-GUI too, but just to prove I could make it appliance-worthy). I'd be interested in any feedback on how to improve it, specifically on how to make Debian and Asterisk take up less space so I could buy the model that only has 512 MB of flash rather than 1 GB. Here's the link. http://tinyurl.com/2hf2cu Let me know what you think. http://www.voip-info.org/files/Embedded_Asterisk.doc There are a few tips in here ofr trimming down debian and having a re-producible build environment. Good luck. JR -- JR Richardson Engineering for the Masses ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multitenant or Multiple virtual machines
I know there are many people using single-tenant and multi-tenant versions of asterisk management and billing packages, but I don't really know if anyone is using it within virtual machines and how well that scales. We have a few FreePBX setups running in virtual machines in environments where the client wants their own PBX (and web interface to play with) without wanting to pay full whack for the server plus hosting, etc. However, we haven't scaled it beyond 3 or 4 VMs per machine - certainly not up to the 10 or 20 VMs you'd be looking at for what you're trying to do. We do, however, have many asterisk installs with 30+ client companies using the same server. In these, each box only runs asterisk once. Each client has SIP identities as follows: companyA-201 companyA-202 companyB-201 etc. Each company has their own context to isolate their extensions from other users. This is definitely a more productive use of resources in that you haven't got the overheads of an OS and asterisk for each client. I wouldn't mind trying it, but since it will probably involve having to purchase multiple software licenses, we want to avoid the $ expense What expense? If you've already got the servers, everything else you need is available open-source. Linux is open source, asterisk is open source, Xen is open source. Don't quote me on this - but isn't there even an open source version of VMware server these days? (There's certainly a free version of VMware server, so even if it isn't open source, there isn't any $ expense) Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
